[asterisk-commits] mnicholson: branch 1.4 r221360 - in /branches/1.4: channels/ configs/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 30 14:36:14 CDT 2009
Author: mnicholson
Date: Wed Sep 30 14:36:06 2009
New Revision: 221360
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=221360
Log:
Fix SRV lookup and Request-URI generation in chan_sip.
This patch adds a new field "portinuri" to the sip dialog struct and the sip peer struct. That field is used during RURI generation to determine if the port should be included in the RURI. It is also used in some places to determine if an SRV lookup should occur.
(closes issue #14418)
Reported by: klaus3000
Tested by: klaus3000, mnicholson
Review: https://reviewboard.asterisk.org/r/369/
Modified:
branches/1.4/channels/chan_sip.c
branches/1.4/configs/sip.conf.sample
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=221360&r1=221359&r2=221360
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Sep 30 14:36:06 2009
@@ -991,6 +991,7 @@
char tag[11]; /*!< Our tag for this session */
int sessionid; /*!< SDP Session ID */
int sessionversion; /*!< SDP Session Version */
+ int portinuri:1; /*!< Non zero if a port has been specified, will also disable srv lookups */
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
@@ -1161,6 +1162,7 @@
ast_group_t pickupgroup; /*!< Pickup group */
struct sockaddr_in addr; /*!< IP address of peer */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
+ int portinuri; /*!< Whether the port should be included in the URI */
/* Qualification */
struct sip_pvt *call; /*!< Call pointer */
@@ -3006,6 +3008,8 @@
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
dialog->maxcallbitrate = peer->maxcallbitrate;
+ if (!dialog->portinuri)
+ dialog->portinuri = peer->portinuri;
return 0;
}
@@ -3025,8 +3029,10 @@
ast_copy_string(peer, opeer, sizeof(peer));
port = strchr(peer, ':');
- if (port)
+ if (port) {
*port++ = '\0';
+ dialog->portinuri = 1;
+ }
dialog->sa.sin_family = AF_INET;
dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
p = find_peer(peer, NULL, 1, 0);
@@ -7419,7 +7425,7 @@
ast_build_string(&invite, &invite_max, "%s@", n);
}
ast_build_string(&invite, &invite_max, "%s", p->tohost);
- if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT)
+ if (p->portinuri)
ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port));
ast_build_string(&invite, &invite_max, "%s", urioptions);
}
@@ -8330,6 +8336,7 @@
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
register_peer_exten(peer, FALSE); /* Remove regexten */
peer->expire = -1;
+ peer->portinuri = 0;
ast_device_state_changed("SIP/%s", peer->name);
/* Do we need to release this peer from memory?
@@ -8583,6 +8590,7 @@
peer->useragent[0] = '\0';
peer->sipoptions = 0;
peer->lastms = 0;
+ peer->portinuri = 0;
pvt->expiry = 0;
if (option_verbose > 2)
@@ -8616,8 +8624,11 @@
if (pt) {
*pt++ = '\0';
port = atoi(pt);
- } else
+ peer->portinuri = 1;
+ } else {
port = STANDARD_SIP_PORT;
+ peer->portinuri = 0;
+ }
oldsin = peer->addr;
/* Check that they're allowed to register at this IP */
@@ -17780,6 +17791,9 @@
/* XXX should unregister ? */
}
+ if (found)
+ peer->portinuri = 0;
+
/* If we have realm authentication information, remove them (reload) */
clear_realm_authentication(peer->auth);
peer->auth = NULL;
@@ -17858,7 +17872,7 @@
ASTOBJ_UNREF(peer_ptr, sip_destroy_peer);
}
ast_clear_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC);
- if (!obproxyfound || !strcasecmp(v->name, "outboundproxy")) {
+ if (!strcasecmp(v->name, "outboundproxy")) {
if (ast_get_ip_or_srv(&peer->addr, v->value, srvlookup ? "_sip._udp" : NULL)) {
ast_log(LOG_ERROR, "srvlookup failed for outboundproxy: %s, on peer %s, removing peer\n", v->value, peer->name);
ASTOBJ_UNREF(peer, sip_destroy_peer);
@@ -17886,6 +17900,7 @@
} else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
peer->contactha = ast_append_ha(v->name + 7, v->value, peer->contactha);
} else if (!strcasecmp(v->name, "port")) {
+ peer->portinuri = 1;
if (!realtime && ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC))
peer->defaddr.sin_port = htons(atoi(v->value));
else
@@ -18006,6 +18021,14 @@
* specified, use that address instead. */
if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) || !peer->addr.sin_addr.s_addr) {
__set_address_from_contact(fullcontact, &peer->addr);
+ }
+ }
+
+ if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC) && !obproxyfound && !ast_strlen_zero(peer->tohost)) {
+ if (ast_get_ip_or_srv(&peer->addr, peer->tohost, srvlookup && !peer->portinuri ? "_sip._udp" : NULL)) {
+ ast_log(LOG_ERROR, "host lookup failed for %s, on peer %s, removing peer\n", peer->tohost, peer->name);
+ ASTOBJ_UNREF(peer, sip_destroy_peer);
+ return NULL;
}
}
Modified: branches/1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/configs/sip.conf.sample?view=diff&rev=221360&r1=221359&r2=221360
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Wed Sep 30 14:36:06 2009
@@ -62,6 +62,9 @@
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
+ ; Specifying a port in a SIP peer definition or
+ ; when dialing outbound calls will supress SRV
+ ; lookups for that peer or call.
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
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